blob: 5fa60dc993ea3e6574bc49eb56132031ccdb1098 [file] [log] [blame]
/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
#include <algorithm>
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
namespace webrtc {
namespace test {
NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
rtc::Optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
return packet_
? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
: rtc::Optional<int64_t>();
}
rtc::Optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
return packet_ ? rtc::Optional<RTPHeader>(packet_->header())
: rtc::Optional<RTPHeader>();
}
void NetEqPacketSourceInput::LoadNextPacket() {
packet_ = source()->NextPacket();
}
std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
if (!packet_) {
return std::unique_ptr<PacketData>();
}
std::unique_ptr<PacketData> packet_data(new PacketData);
packet_->ConvertHeader(&packet_data->header);
if (packet_->payload_length_bytes() == 0 &&
packet_->virtual_payload_length_bytes() > 0) {
// This is a header-only "dummy" packet. Set the payload to all zeros, with
// length according to the virtual length.
packet_data->payload.SetSize(packet_->virtual_payload_length_bytes());
std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
} else {
packet_data->payload.SetData(packet_->payload(),
packet_->payload_length_bytes());
}
packet_data->time_ms = packet_->time_ms();
LoadNextPacket();
return packet_data;
}
NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name,
const RtpHeaderExtensionMap& hdr_ext_map)
: source_(RtpFileSource::Create(file_name)) {
for (const auto& ext_pair : hdr_ext_map) {
source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
}
LoadNextPacket();
}
rtc::Optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
return next_output_event_ms_;
}
void NetEqRtpDumpInput::AdvanceOutputEvent() {
if (next_output_event_ms_) {
*next_output_event_ms_ += kOutputPeriodMs;
}
if (!NextPacketTime()) {
next_output_event_ms_ = rtc::Optional<int64_t>();
}
}
PacketSource* NetEqRtpDumpInput::source() {
return source_.get();
}
NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name,
const RtpHeaderExtensionMap& hdr_ext_map)
: source_(RtcEventLogSource::Create(file_name)) {
for (const auto& ext_pair : hdr_ext_map) {
source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
}
LoadNextPacket();
AdvanceOutputEvent();
}
rtc::Optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
return rtc::Optional<int64_t>(next_output_event_ms_);
}
void NetEqEventLogInput::AdvanceOutputEvent() {
next_output_event_ms_ =
rtc::Optional<int64_t>(source_->NextAudioOutputEventMs());
if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
next_output_event_ms_ = rtc::Optional<int64_t>();
}
}
PacketSource* NetEqEventLogInput::source() {
return source_.get();
}
} // namespace test
} // namespace webrtc